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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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179 // - payload_type : the payload-type to be removed. | 179 // - payload_type : the payload-type to be removed. |
180 // | 180 // |
181 // Return value : 0 if OK. | 181 // Return value : 0 if OK. |
182 // -1 if an error occurred. | 182 // -1 if an error occurred. |
183 // | 183 // |
184 int RemoveCodec(uint8_t payload_type); | 184 int RemoveCodec(uint8_t payload_type); |
185 | 185 |
186 // | 186 // |
187 // Remove all registered codecs. | 187 // Remove all registered codecs. |
188 // | 188 // |
189 void RemoveAllCodecs(); | 189 int RemoveAllCodecs(); |
190 | 190 |
191 // Returns the RTP timestamp for the last sample delivered by GetAudio(). | 191 // Returns the RTP timestamp for the last sample delivered by GetAudio(). |
192 // The return value will be empty if no valid timestamp is available. | 192 // The return value will be empty if no valid timestamp is available. |
193 rtc::Optional<uint32_t> GetPlayoutTimestamp(); | 193 rtc::Optional<uint32_t> GetPlayoutTimestamp(); |
194 | 194 |
195 // Returns the current total delay from NetEq (packet buffer and sync buffer) | 195 // Returns the current total delay from NetEq (packet buffer and sync buffer) |
196 // in ms, with smoothing applied to even out short-time fluctuations due to | 196 // in ms, with smoothing applied to even out short-time fluctuations due to |
197 // jitter. The packet buffer part of the delay is not updated during DTX/CNG | 197 // jitter. The packet buffer part of the delay is not updated during DTX/CNG |
198 // periods. | 198 // periods. |
199 // | 199 // |
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279 Clock* clock_; // TODO(henrik.lundin) Make const if possible. | 279 Clock* clock_; // TODO(henrik.lundin) Make const if possible. |
280 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); | 280 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); |
281 rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_); | 281 rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_); |
282 }; | 282 }; |
283 | 283 |
284 } // namespace acm2 | 284 } // namespace acm2 |
285 | 285 |
286 } // namespace webrtc | 286 } // namespace webrtc |
287 | 287 |
288 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ | 288 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ |
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